A. Field of the Invention
The present invention relates to a signal processing method to compress digital audio data by a block floating process and, more particularly, to digital signal encoding/decoding apparatuses suitable for preventing deterioration of the sound quality of a digital audio signal and for improving a compression ratio of the digital audio signal when the digital audio signal is converted on a frequency axis per predetermined time frame, and data on the frequency axis is then divided into a plurality of blocks to be recorded, otherwise, bits are then allocated per block to compress the digital audio signal for minimizing transmitted parameters, and encoding/decoding methods thereof.
B. Description of the Related Art
High efficiency methods of coding digital audio signals include a subband coding (hereinafter referred to as "SBC") method that divides a signal on the time axis of an audio signal into a plurality of frequency bands to code the divided signals, as well as an adaptive transform coding (ATC) method that vertically transforms a signal on a time axis to a signal on a frequency axis to divide the signal into a plurality of frequency bands and adaptively code to respective bands.
FIG. 1 is a block diagram showing a conventional digital data encoder adopting the adaptive transform coding method. As shown in FIG. 1, the conventional digital data encoder includes a PCM data block circuit 1 for dividing received spectrum data PCM into sub-blocks on several time axes. A band-split filter 2 divides output data from the PCM data block circuit 1 per predetermined frequency as in SBC. Furthermore, a time/frequency converter 3 receives output data per frequency band from the band-split filter 2 to convert the received data into a frequency axis perpendicular to the time axis, thereby outputting spectrum data. A scale factor operator 4 allows the spectrum data from the time/frequency converter 3 to be blocks for obtaining scale factor SF per block. To calculate a minimum masking level per block, a masking level operator 5 searches out a tonal component of the spectrum data, i.e., spectrum data of a specific sound, from the time/frequency converter 3 and obtains a masking level. A bit allocation operator 6 determines a word length WL output to allocate bits per block for generating a quantization error smaller than the masking level from the masking level operator 5. A parameter operator 7 obtains parameters by means of a signal from the masking level operator 5 and bit allocation operator 6. A spectrum signal normalizer and quantizer 8 produces quantized spectrum by means of the scale factor SF from the scale factor operator 4 and word length WL from the bit allocation operation 6.
The operation of the conventional digital data encoder constructed as above will be described below.
The PCM audio data on the time axis is formed to be a block through the PCM data block circuit 1, and then band-divided into the predetermined number of subbands (e.g., three or four) via the band-split filter 2 which is similar to the SBC. The subbands are supplied to the time/frequency converter 3. In the time/frequency converter 3, the subbands are subjected to fast Fourier transform (FFT) or discrete cosine transform (DCT) to become a FFT coefficient (or DCT coefficient) that is the spectrum data.
As shown in FIG. 2, the FFT coefficient from the time/frequency converter 3 is divided into blocks B.sub.0, B.sub.1, . . . , B.sub.n-1, and then the scale factor SF per block is obtained in the scale factor operator 4. At this time, the scale factor SF may use a peak value of each block or a value obtained by multiplying the average value by a specific coefficient.
A floating coefficient of each block is calculated through the scale operator 4, so that respective blocks are normalized with the floating coefficient, and quantized to the number of bits obtained in the spectrum signal normalizer and quantizer 8. In this case, either the peak value of each block or the value obtained by multiplying the average value by a prescribed coefficient are used as the floating coefficient; otherwise, the floating coefficient is quantized, and the quantized floating coefficient is named as the scale factor SF. Meanwhile, the masking level converter 5 searches out the tonal component of the FFT coefficient (or DCT coefficient) from the time/frequency converter 3 to obtain the masking level considering a man's audible characteristic, and obtains the minimum masking value per block.
Thereafter, the bit allocation operator 6 allocates bits to produce a quantization error smaller than the masking level in accordance with the masking level per block. In other words, the bit allocation operator 6 determines the word lengths WL of respective blocks. Then, the parameter operator 7 receives the outputs from the masking level operator 5 and the bit allocation operator 6 to operate the number of blocks recorded or transmitted. That is, the parameter operator 7 determines the number N of the sub-blocks recorded or transmitted.
The spectrum signal normalizer and quantizer 8 normalizes the spectrum signal for each block with respect to the scale factor SF obtained in the scale factor operator 4, and quantizes the spectrum signal in accordance with the word length WL determined in the bit allocation operator 6.
The digital audio data is compressed as described above, and the information, such as the scale factor SF from the scale factor operator 4, word length WL from the bit allocation operator 6, the number N of blocks from the parameter operator 7, as well as quantization data QSP of the spectrum signal from the spectrum signal normalizer and quantizer 8, are recorded or transmitted. Here, the scale factor SF, word length WL and the number N of blocks are designated as side information which is requisite in the conventional technique.
When the conventional ATC method, that necessarily records or transmits three pieces of side information, is applied in the conventional digital signal processing system, there is no problem of favorably executing the compression of the digital audio signal in a certain ratio. Where the parameter operator 7 decreases the number N of blocks in case of unfavorable compression, however, information, is lost when restoring the information thus deteriorating sound quality.